Index: webrtc/call/bitrate_estimator_tests.cc |
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc |
index 80949baba4d5933d672511385076e4c8b7ae5560..ff72c6688a61a20f38c31c15167281b2e95ba21c 100644 |
--- a/webrtc/call/bitrate_estimator_tests.cc |
+++ b/webrtc/call/bitrate_estimator_tests.cc |
@@ -18,6 +18,7 @@ |
#include "webrtc/base/logging.h" |
#include "webrtc/base/thread_annotations.h" |
#include "webrtc/call.h" |
+#include "webrtc/modules/audio_device/include/mock_audio_device.h" |
#include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
#include "webrtc/system_wrappers/include/trace.h" |
#include "webrtc/test/call_test.h" |
@@ -109,6 +110,8 @@ class BitrateEstimatorTest : public test::CallTest { |
virtual void SetUp() { |
AudioState::Config audio_state_config; |
audio_state_config.voice_engine = &mock_voice_engine_; |
+ ON_CALL(mock_voice_engine_, audio_device_module()) |
+ .WillByDefault(Return(&mock_audio_device_)); |
stefan-webrtc
2016/11/03 14:33:53
Could you try to remove everything audio related h
aleloi
2016/11/08 11:46:41
I've done it in another CL; I'll rebase this one a
|
Call::Config config(&event_log_); |
config.audio_state = AudioState::Create(audio_state_config); |
receiver_call_.reset(Call::Create(config)); |
@@ -252,6 +255,7 @@ class BitrateEstimatorTest : public test::CallTest { |
}; |
testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_; |
+ testing::NiceMock<webrtc::test::MockAudioDeviceModule> mock_audio_device_; |
LogObserver receiver_log_; |
std::unique_ptr<test::DirectTransport> send_transport_; |
std::unique_ptr<test::DirectTransport> receive_transport_; |